search for: gandikota

Hi All Has anyone used Asterisk for a Call Center operation? What I mean is: given a list of phone numbers, can Asterisk dial each number, play a message and accept some DTMF? I ask because I am an employee of a non-profit company based in San Diego, CA. I already evaluated Voicent and Voxeo. The former has expensive licensing terms and the latter is not best suited for a call center. I would

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous",...

On Thu, 6 Aug 2015, Murthy Gandikota wrote: [trimming cruft nobody cares about anymore] > I use the same password for INBOUND and it works fine! Something amiss > with Asterisk OUTBOUND? because I used the same password with X-Lite and > X-Pro Vonage soft phones with successful calls. Would comparing an INVITE from X-Lit...

...ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Murthy Gandikota Sent: Saturday, December 06, 2014 8:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Playing audio to bridged channels I would like to play audio--using controlplayback-- to 2 channels--agent and caller- simultaneously. Tried meetme,confbridge,originate...

On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous",...

On Thu, 6 Aug 2015, Steve Edwards wrote: > Would comparing an INVITE from X-Lite or X-Pro with the INVITE from > Asterisk yield any clues? On Thu, 6 Aug 2015, Murthy Gandikota wrote: > For Asterisk INVITE please view > > http://pastebin.com/v15vMax4 > > For X-Lite INVITE please view > > http://pastebin.com/rmHZKu3N Just a quick glance (because I'm not a SIP expert)... (Asterisk) Request-Line: INVITE sip:1732xxxxxxx at 69.59.234.67:5061 SIP/2....

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > &...

...oup is pretty much dead, so if you run into a problem, there's really nowhere to go. Overall, I think this is a good application. I hope this helps. Regards; John V. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Murthy Gandikota Sent: Monday, December 08, 2014 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Playing audio to bridged channels using ControlPlayBack There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html...

On Wed, Jul 29, 2015 at 10:16 AM, John Novack <jnovack at stromberg-carlson.org > wrote: > > > Murthy Gandikota wrote: > > > > ------------------------------ > To: asterisk-users at lists.digium.com > From: webaccounts173 at jgoettgens.de > Date: Wed, 29 Jul 2015 16:11:31 +0200 > Subject: Re: [asterisk-users] Windows Asterisk Help > > > > Downloaded latest version of Aste...

To: asterisk-users at lists.digium.com From: webaccounts173 at jgoettgens.de Date: Wed, 29 Jul 2015 16:11:31 +0200 Subject: Re: [asterisk-users] Windows Asterisk Help Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf

Hi All I am trying to dial out using SIP and Vonage using the instructions : <a href="http&#58;&#47;&#47;www.voip-info.org&#47;wiki&#47;view&#47;Asterisk&#43;and&#43;Vonage" target="_blank"

Hi All, Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV

Hi masters, I?m not an expert on this my friends, but I?m trying to understand which the expected behaviour is from Asterisk side when you deal with the following scenario: Caller ?> GSM Gateway with SIM card A ?> Asterisk queue ?> extension 1000 GSM gateway with call waiting activated on SIM A Queue with ?skip busy agent? disabled and ringall strategy. SIP extension 1000 with call

I would like to play audio--using controlplayback-- to 2 channels--agent and caller- simultaneously. Tried meetme,confbridge,originate without success. Tried redirecting the channels to a context, playing audio to the agent's channel and then bridging the 2 channels. The problem with this is as soon as the bridge is created the audio stops. I can provide the dialplan details, if anyone is

Hi All My dial plan has the following context: [sip-guest] exten => _!.,1, Answer exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}]) exten => _!.,n, resetcdr(w) exten => _!.,n, resetcdr(w) exten => _!.,n, set(DNIS=${EXTEN}) exten => _!.,n, resetcdr(w) exten => _!.,n,

I keep getting the following message whenever an AMI call is made: asteriskjava.manager.internal.EventBuilderImpl.buildEvent(EventBuilderIm pl.java:296) No event class registered for event type 'localbridge', Tried adding an event listener. Anyone know how to fix this? Thanks Murthy -------------- next part -------------- An HTML attachment was scrubbed... URL: